similar to: Queue Monitor

I created some scripts to logrotate. I am having a problem. After I do it, I am sending kill -HUP to the process its not using the newly created messages file again. Could someone help me out with how I can rotate asterisk's log's without killing the process? ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL

Im having some problems with connecting a TNT to asterisk. The problem is when the call is sent to asterisk and signaling is done the RTP syncs however no audio is produced. Can someone give me some idea of how to accomplish this? I am using the standard configs and g711 and 729 do the same. No audio. Public IPs on both ends. No nat. Any ideas would be appreciated.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello I am running an * box with just 8 extensions connected to our old Alcatel BCN 5200 PABX. The requirement is that we now scale it up to handle about 300 lines and get rid of our old PABX. Is there a way of hooking up 300 phones to asterisk without going via the PABX. I am more of a network person than a telecomms one so i may not fully

I currently have a single server with a few SIP and IAX upstreams for origination and termination with IAX clients. I am adding a second server that will have a much higher capacity and will be handling a larger call volume. However, this second server is not going to be geographically near the first. It will largely share the same upstreams. I would like for this to be an integrated system

Does anyone know of any companies where I can interconnect with for SMS? ? .o-------------------------------------------------------o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office

Can someone explain what this error is? -- Got SIP response 500 "Server Internal Error - Invalid CSEQ number" back from 209.xxx.xxx.xxx How do I fix this? .o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office

Does anyone in here know how to setup auto negotiation between g729 and g711ulaw on a cisco 5400? I would imagine it would be the same on a 3660. The problem I am having is natively the call is setup for g729 however when the call is transferred to voicemail it uses ULAW so when the cisco tries to connect to the voice mail I get a SIP error that the codec couldn't be negotiated. I need

are you giving answer()? ..o-------------------------------------------------------o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users

I had the same problem and I believe it was the payload size of the codec. What code are you using? ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Federico Alves Sent: Friday,

I don't completely understand this.. Lemme try it out.. [default] exten => 1112223333,1,Macro(happy-did) [macro-happy-did] exten => s,1,Goto(${MACRO_EXTEN},1) exten => _XXXXXXXXXX,1,NoOp(Normal "s" exten stuff here) So when this is ran it will cut the cdr and the s will show the actual DID not the s correct? But then the NoOp would be something like: ....

I run a bunch of the Linksys ATA's.. I always use rfc2833 for DTMF. works very well and have never had a problem with it. ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

To begin with, I am new to both asterisk and VOIP and although I've gotten pretty far with my Asterisk setup and have two different sip accounts working fine for outgoing calls I am having trouble with one issue. My problem is that I have another provider who uses IAX2 that I wish to use for calling various countries, including local (The Netherlands) calls and calls to the US to

Hello all, We are looking for some hardware requirements/recommendations to be able to handle a full DS3's worth of TDM -> VoIP traffic. The DS3 would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to convert those calls into G729 SIP VoIP calls to send to our asterisk box over ethernet. Since everything is going in/out of asterisk is 729, and no features

I have delete=yes and attach=yes. But my messages are not getting deleted after they're sent. I'm running asterisk as root so it can't be a permission issue. Any ideas?

Does anybody out there have any evidence that Sipura support is still in operation? rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed.

Running fc3 with current cvs-head... Is there a nice way to rotate the /var/log/asterisk/messages file without shutting down asterisk? I'm currently rotating the log files via cron, however my script requires asterisk to be shut down, which also kills any outstanding cli sessions (eg, asterisk -rvvvvv). Would like to rotate the files without killing the cli session. Any reasonable way to

Hi, I have set up a Samba 2.2.7a to work as a PDC for my Win2k boxes. They can join the domain suplied by samba, and all the users can log on to their domain but, the roaming profiles just won't work. I'm using win2k SP4 and my smb.conf is as described bellow: [global] workgroup = CASA netbios name = SERVIDOR interfaces = 192.168.0.1/255.255.255.0 127.0.0.1/255.0.0.0 bind interfaces

don't quote me on this but I believe the earlier assumtion is correction. I think you need to have RTP going bothways otherwise the call will disconnect. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Chris Shaw Sent: Friday, August 13, 2004 12:40 PM To: asterisk-users@lists.digium.com Subject: Re:

> -----Original Message----- > From: Asterisk@gtcus.com [mailto:Asterisk@gtcus.com] > Sent: Saturday, September 06, 2003 8:39 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] VONAGE or IP Dialtone > > > Thanks for the great feedback on these options. I am fairly > new at this and not familiar with the IAX/IAX2 capabilities > offered by

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